The ever-growing traffic load in communication networks has been realized for quite some time as a problem that requires a solution. To date, some solutions were suggested to the problem. One such solution was suggested in US 20030012137 that describes a packet network congestion control system using a biased packet discard policy. Once a connection and session are established, compressed voice and data packets start flowing between the two end points of the path. A control entity supplies congestion control packets periodically. The control packets provide a “heartbeat” signal to the codec at the other end of the session. Each codec receiver uses the “heartbeat” signal as an processing indication of network congestion. As network congestion increases, routers within the network discard excess packets to prevent network failure. The network discards all packets classified as congestion control packets whenever a flow control mechanism detects congestion or a trend toward congestion. As packets are discarded, the end points renegotiate codec type and/or parameters to realize lower bit rates.
WO 0057606 describes a method for discarding data within an IP-network communications link. Initially, the IP-network communications link is monitored to determine the occurrence of an overload condition. At least some of the data packets transmitted along the IP-network communications link are selected in response to detection of the overload condition, and the selected data packets are discarded from the link, while the remainder of the packets are transmitted. When packets of real-time flow sessions are concerned, the selected data packets are those having the same source and destination IP addresses and source and destination ports, and consisting of encapsulated video, audio, etc. signals. Discarding these packets from that link allows that only a single or a few real-time flow sessions are eliminated from the link, while the remaining links' sessions are unaffected.
U.S. Pat. No. 6,091,709 discloses a packet router which is provided with priority services of the type required for isochronous handling of data representing real-time voice, includes a Quality of Service (QoS) management system for ensuring that guarantees associated with such priority service can be met with a high degree of certainty. This management system provides prioritized queues including a highest priority queue supporting reservations for the priority service suited to isochronous handling. The highest priority queue and other queues are closely monitored by a QoS manager element for states of near congestion and critical congestion. While neither state exists, filler packet flows are promoted from lower priority queues to the highest priority queue, in order to keep the latter queue optimally utilized. If all lower priority queues are empty at such times, dummy packets which will be discarded by stations receiving them, are inserted as filler flows. When a state of near congestion exists, the QoS manager demotes filler flow units from the highest priority queues to lower priority queues, in order to lessen the potential forwarding delays presented to real traffic occupying the highest priority queue. When a state of critical congestion exists in the highest priority queue, admission of new incoming traffic flows to that queue is suspended and forwarding of filler flows from that queue out to the network is also suspended.
The following publications were published as Requests For Comments (RFCs) by the Internet Society for the establishment of an international standard DiffServ: RFC 2474 “Definition of the Differentiated Services Field (DS Field) in the Ipv4 and Ipv6 Headers”, by K. Nichols et al., December 1998; RFC 2475 “An Architecture for Differentiated Services”, by S. Blake et al., December 1998; RFC 2597 “Assured Forwarding PHB Group”. By J. Heinanen et al., June 1999; RFC 2836 “Per Hop Behavior Identification Codes”, by D. Black et al. June 2001; and RFC 3260 “New Terminology and Clarifications for Diffserv” by D. Grossman, April 2002.
However, as may be appreciated, these solutions are either directed to solve overload problems or to ensure quality of service. However, none of these publications disclose how to carry out different processing of various packets (or their parts, if applicable) belonging to the same signal, so as to allow optimizing the network resources.
In our co-pending application, IL 160921, a method for managing varying traffic load in a packet switched communication network is disclosed. By this method the active channels carrying traffic are divided into groups and a rate adjusting mechanism is applied thereon when the available bandwidth is less than the bandwidth required. The application of the rate adjusting mechanism is done while ensuring that a substantially equalized signal quality is maintained for traffic delivered via all of the active channels belonging to certain group(s).
Still, even this solution does have certain drawbacks as it is more suitable for actions to be taken near a point where the rate adjustment is applied on the delivered traffic, and does not necessarily provide a solution that ensures certain quality to certain channels and/or signals further downstream.
The disclosures of all references mentioned above and throughout the present specification are hereby incorporated herein by reference.